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The extensions I've created in AAH, when dialed, always go straight to
voicemail. I may be missing a step... I'm simply adding it in the "Extensions" part of AAH. I can dial out with my extension, and recieve the voicemail notification, so I know i'm logged in, or so I thought... This is SIP 210 logging in and 220 making a call to 210 ------------------- asterisk1*CLI> Sip read: REGISTER sip:192.168.0.50 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Max-Forwards: 70 Contact: Jeremi Bergman <sip:[EMAIL PROTECTED]:5060>;expires=3600 User-Agent: Sipura/SPA3000-2.0.10(GWc) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 12 headers, 0 lines Using latest request as basis request Sending to 192.168.0.8 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=as0373da7c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 asterisk1*CLI> to 192.168.0.8:5060 Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=as0373da7c Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> WWW-Authenticate: Digest realm="asterisk", nonce="04ab00ad" Content-Length: 0 to 192.168.0.8:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk1*CLI> Sip read: REGISTER sip:192.168.0.50 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER Max-Forwards: 70 Authorization: Digest username="210",realm="asterisk",nonce="04ab00ad",uri="sip:[EMAIL PROTECTED]",algorithm=MD5,response="4b5484b65bc24fc38c8cdff7684a9452"; Contact: Jeremi Bergman <sip:[EMAIL PROTECTED]:5060>;expires=3600 User-Agent: Sipura/SPA3000-2.0.10(GWc) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura 13 headers, 0 lines Using latest request as basis request Sending to 192.168.0.8 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=as0373da7c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.8:5060 Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=49a471fb8d817603o0 To: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=as0373da7c Call-ID: [EMAIL PROTECTED] CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Expires: 3600 Contact: <sip:[EMAIL PROTECTED]:5060>;expires=3600 Date: Wed, 06 Jul 2005 15:53:45 GMT Content-Length: 0 to 192.168.0.8:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms 11 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as60a8b668 To: <sip:[EMAIL PROTECTED]:5060> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 42 Messages-Waiting: no Voice-Message: 0/0 (no NAT) to 192.168.0.8:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms asterisk1*CLI> Sip read: SIP/2.0 200 OK To: <sip:[EMAIL PROTECTED]:5060>;tag=e365bb5ba561bc23i0 From: "Unknown" <sip:[EMAIL PROTECTED]>;tag=as60a8b668 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 Server: Sipura/SPA3000-2.0.10(GWc) Content-Length: 0 8 headers, 0 lines Destroying call '[EMAIL PROTECTED]' Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> asterisk1*CLI> Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 44887 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 292 v=0 o=220 782441328 782441343 IN IP4 192.168.0.2 s=X-Lite c=IN IP4 192.168.0.2 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 11 headers, 13 lines Using latest request as basis request Sending to 192.168.0.2 : 5060 (non-NAT) Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as063c6cef Call-ID: [EMAIL PROTECTED] CSeq: 44887 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="71694d20" Content-Length: 0 to 192.168.0.2:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Found user '220' asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as063c6cef Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 44887 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]> Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 44888 INVITE Proxy-Authorization: Digest username="220",realm="asterisk",nonce="71694d20",response="d1ee9ed08d97c4b058f97a24f64ba5a4",uri="sip:[EMAIL PROTECTED]"; Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 292 v=0 o=220 782441328 782441343 IN IP4 192.168.0.2 s=X-Lite c=IN IP4 192.168.0.2 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 13 lines Using latest request as basis request Sending to 192.168.0.2 : 5060 (non-NAT) Found user '220' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.2:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Looking for 210 in from-internal list_route: hop: <sip:[EMAIL PROTECTED]:5060> Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as3714a6d8 Call-ID: [EMAIL PROTECTED] CSeq: 44888 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.2:5060 -- Executing Macro("SIP/220-0f69", "exten-vm|[EMAIL PROTECTED]|210";) in new stack -- Executing SetVar("SIP/220-0f69", "FROMCONTEXT=exten-vm") in new stack -- Executing Macro("SIP/220-0f69", "record-enable|210|IN") in new stack -- Executing GotoIf("SIP/220-0f69", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing GotoIf("SIP/220-0f69", "0?5:8") in new stack -- Goto (macro-record-enable,s,8) -- Executing GotoIf("SIP/220-0f69", "0?9:12") in new stack -- Goto (macro-record-enable,s,12) -- Executing DBget("SIP/220-0f69", "RecEnable=RECORD-IN/210") in new stack -- DBget: varname=RecEnable, family=RECORD-IN, key=210 -- DBget: Value not found in database. -- Executing SetVar("SIP/220-0f69", "CALLFILENAME=20050706-115857-1120665537.5") in new stack -- Executing GotoIf("SIP/220-0f69", "0?15:99") in new stack -- Goto (macro-record-enable,s,99) -- Executing NoOp("SIP/220-0f69", "NO RECORDING NEEDED") in new stack -- Executing Macro("SIP/220-0f69", "dial|15|tr|210") in new stack -- Executing GotoIf("SIP/220-0f69", "0?4:2") in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf("SIP/220-0f69", "0?4:3") in new stack -- Goto (macro-dial,s,3) -- Executing SetCIDName("SIP/220-0f69", "David Johnson") in new stack -- Executing AGI("SIP/220-0f69", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 -- Executing NoOp("SIP/220-0f69", "Returned from dialparties with no extensions to call") in new stack -- Executing SetVar("SIP/220-0f69", "DIALSTATUS=BUSY") in new stack -- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new stack -- Executing GotoIf("SIP/220-0f69", "0?s-BUSY|1") in new stack -- Executing NoOp("SIP/220-0f69", "Sending to Voicemail box [EMAIL PROTECTED]";) in new stack -- Executing Macro("SIP/220-0f69", "vm|[EMAIL PROTECTED]|BUSY";) in new stack -- Executing Goto("SIP/220-0f69", "s-BUSY|1") in new stack -- Goto (macro-vm,s-BUSY,1) -- Executing VoiceMail("SIP/220-0f69", "[EMAIL PROTECTED]";) in new stack We're at 192.168.0.50 port 17334 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK2A632B189EC04A0CBAEB17D1B3342931 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as3714a6d8 Call-ID: [EMAIL PROTECTED] CSeq: 44888 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Type: application/sdp Content-Length: 238 v=0 o=root 1983 1983 IN IP4 192.168.0.50 s=session c=IN IP4 192.168.0.50 t=0 0 m=audio 17334 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.0.2:5060 -- Playing 'vm-theperson' (language 'en') asterisk1*CLI> Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK39CF2EA3F9794B68A722E58AB6B681D1 From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as3714a6d8 Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 44888 ACK Max-Forwards: 70 Content-Length: 0 9 headers, 0 lines -- Playing 'digits/2' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'vm-isonphone' (language 'en') asterisk1*CLI> Sip read: 0 headers, 0 lines -- Playing 'vm-intro' (language 'en') -- Playing 'beep' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav49, 0x8615f78 -- x=1, open writing: /var/spool/asterisk/voicemail/default/210/INBOX/msg0001 format: wav, 0x8605ea0 asterisk1*CLI> Sip read: BYE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as3714a6d8 Contact: <sip:[EMAIL PROTECTED]:5060> Call-ID: [EMAIL PROTECTED] CSeq: 44889 BYE Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 10 headers, 0 lines Sending to 192.168.0.2 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bKBB5910FF6B464AA2AC22A34DAC7CE76F From: Jeremi Bergman <sip:[EMAIL PROTECTED]>;tag=2051996763 To: <sip:[EMAIL PROTECTED]>;tag=as3714a6d8 Call-ID: [EMAIL PROTECTED] CSeq: 44889 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.2:5060 -- User hung up -- Recording was 2 seconds long but needs to be at least 3 - abandoning == Spawn extension (macro-vm, s-BUSY, 1) exited non-zero on 'SIP/220-0f69' in macro 'vm' == Spawn extension (macro-exten-vm, s, 7) exited non-zero on 'SIP/220-0f69' in macro 'exten-vm' == Spawn extension (from-internal, 210, 1) exited non-zero on 'SIP/220-0f69' -- Executing Macro("SIP/220-0f69", "hangupcall") in new stack -- Executing ResetCDR("SIP/220-0f69", "w") in new stack -- Executing NoCDR("SIP/220-0f69", "") in new stack -- Executing Wait("SIP/220-0f69", "5") in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/220-0f69' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/220-0f69' Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> --------------- Thanks in advanced! |
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